2f71b61a34
In refactoring CL https://webrtc-review.googlesource.com/c/src/+/210340, the RTCRemoteInboundRtpStreamStats hierarchy was updated to inherit from RTCReceivedRtpStreamStats but we forgot to update the WEBRTC_RTCSTATS_IMPL() macro to say that RTCReceivedRtpStreamStats is the parent. As a consequence, RTCReceivedRtpStreamStats's members (jitter and packetsLost) were not included when iterating over all members of RTCRemoteInboundRtpStreamStats, which means these two merics stopped being exposed to JavaScript in Chromium. There is sadly no way to safe-guard against this, but the fix is simple. TBR=hta@webrtc.org,meetwudi@gmail.com Bug: webrtc:12532 Change-Id: I0179dad6eaa592ee36cfe48978f2fc22133b8f45 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212866 Reviewed-by: Henrik Boström <hbos@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33543}
WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.
Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.
The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.
Development
See here for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://webrtc.googlesource.com/src
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: http://build.chromium.org/p/client.webrtc
- Coding style guide
- Code of conduct
- Reporting bugs